Is there a way to get a report from Asterisk on the quality metrics
(packet loss, delay, jitter) of at least the inbound component of a SIP
call?
Thanks
James
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Hi
Was able to communicate clearly with e60 and E61
with asterisk with new access point , even though the
access point security setting was of opennetworks ,
the previous one was of WEP , I feel this was a
major hurdle in communication , now I can clearly
accept and make calls using
Nick Chalk wrote:
[EMAIL PROTECTED] wrote:
I've got speedtouch ones at home, here I've got
a Zoom one and a Dlink one I can try, It will be
a bit of a botch-job, atm. I'm using one of
those nice ones that plug into the front of an
NTE-5 (so I can punch the cables straight in).
An
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
I
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
no on-hook state in the whole setup.
Obviously If I just connect the input to a port on
Thanks a lot for responding.
I did what you recomended, and it works now. At least I can make simple
calls out. Did not try the incoming part though.
Now it is still unclear :
- how to make the Dial application choose the first available channel?
the easiest (for you) is installing freepbx
Keyboardot ragadtam, hogy va'laszoljak Tigran Kocharyan osszedobalt bytejaira:
1. Customer Calls the outgoing number which is a PSTN line connected to
my Zap channel
2. Asterisk captures the Caller ID and calls back the customer.
3. As soon as the customer picks up the phone, asterisk plays
James Harper wrote:
Easy to do on the Linksys PAP2, if that helps. The functionality
probably depends on the make and model of the phone... maybe if you
gave
those details as well?
James
Fantastic, this may solve the problem In the mail I've just posted
(which hasnt' appeared yet).
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as 'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put it into
Dear list,
I've been looking for a voip service provider with inexpensive and high-
quality call service to China. However, the providers I've tried
(voipjet, exgn, voxee) all have long to super-long latencies on calls to
China.
Has anyone found a service with good connections to China? Please
hi !
i have installed asterisk-1.2.9.1
but am unable to run it
i am getting this error
[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined
symbol: ast_pthread_create
Jun 11 16:43:00 WARNING[8968]: loader.c:554
On Sun, 2006-06-11 at 20:52 +1000, James Harper wrote:
Ideally I would have liked the pap2 to have done the same as
'immediate'
when talking about fxo, capi, misdn, etc, but I couldn't get it to
automatically dial nothing. A '0' was the best I could do. If anyone
knows how to put it into
James Harper wrote:
So... asterisk can't tell the difference between 's' for 'no extension
dialled', and when 's' was actually the name of the extension dialled...
is this the expected behaviour?
I surely hope so, you can refer to it as such in the extensions.conf as
well (with goto etc.)
amna saleem wrote:
hi !
i have installed asterisk-1.2.9.1
but am unable to run it
i am getting this error
[pbx_wilcalu.so]Jun 11 16:43:00 WARNING[8968]: loader.c:325
__load_resource: /usr/lib/asterisk/modules/pbx_wilcalu.so: undefined
symbol: ast_pthread_create
Jun 11 16:43:00
Thomas Kenyon wrote:
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
Does the connection use 2 screws for analog inputs? If this
Doug Lytle wrote:
Thomas Kenyon wrote:
I need to be able to connect an old PA system to an asterisk box, which
basically works as a couple of amplifiers taking an analogue phone
signal and playing whatever it produces out of some speakers. There is
Does the connection use 2 screws for
Are there any known problems with Cisco routers (Cisco 837) and SIP
sessions? I have been trying to track down a problem for about 3 hours
now and I think the Cisco router is the culprit!!!
I keep getting 488 Not acceptable here messages, which are apparently
normally the message you get when a
I am trying to use hook flash to transfer a call but I want the recording
on the line I transfer to to start after I hang up. In other words if I
receive a call and want to transfer it to VM or to a recording, I want to
be able to flash the hook, dial the extension, and hang up. But I do not
want
Additionally, just to satisfy myself that I wasn't going mad I changed
the port from 5060 to 5070 and now things are working, so something is
definitely playing up on port 5060.
James
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of
Destar[1] has recentely included Virtual PBX features inside it's main funcionality (right now you have to download the trunk developement branch to get it), but it would be availabe on version 0.2 coming soon in a few weeks.
[1] http://destar.berlios.de/jmaczOn 6/9/06, William Piper
[EMAIL
I had a bunch of PSP2-NA devices with firmware 3.x that did that.
Downgrading to 2.0.13 solved the problem. Others said that the last
3.x would do also, but after putting out hundreds of PAP2 with 2.x and
they all working rock solid, I'm not willing to switch to 3.x until I
have tested it enough
I guess I've found some good references on how to accomplish this:
http://voxilla.com/PNphpBB2-viewtopic-t-6320-sid-11997b0cebea526d7a7562f38c0fd595.html
http://nerdvittles.com/index.php?p=73
Thanks for the hint though.
Woodoo People .pGa! wrote:
Keyboardot ragadtam, hogy va'laszoljak
Tzafrir Cohen a écrit :
I'm still not hapy with that as a default. It should provide you a basis
for manual editing at this stage. But I wonder what else could the
script configured there differently. Are those sane defaults for BRI on
France?
I've modified zaptel-channels.conf file ,
Tzafrir Cohen a écrit :
Jut as usual with Zaptel: Zap/NNN (e.g: Zap/1 , Zap/2) for individual
channels. And gNNN and similar work just the same.
OK, in extensions.conf, i put the contexts PSTN and INTERNAL as:
[PSTN] ; for in coming calls - defin in zapata.conf
exten =
On Jun 11, 2006, at 2:32 AM, John Joseph wrote:
Hi
Was able to communicate clearly with e60 and E61
with asterisk with new access point , even though the
access point security setting was of “opennetworks” ,
the previous one was of “WEP” , I feel this was a
major hurdle in communication
On Jun 11, 2006, at 8:15 AM, James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I changed
the port from 5060 to 5070 and now things are working, so something is
definitely playing up on port 5060.
If you are behind a NAT perhaps two SIP devices are both trying to
Hi Amna,
Make a test
In the archive modules.conf places the following line: noload = pbx_wilcalu.soStopasterisk and initiates asterisk again. It mustresolv its problem.
I wait to have helped.
Greetings
Josué
2006/6/11, Thomas Kenyon [EMAIL PROTECTED]:
amna saleem wrote: hi ! i have installed
James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I changed
the port from 5060 to 5070 and now things are working, so something is
definitely playing up on port 5060.
James
You probably have are behind NAT and your NAT device has a SIP ALG.
Changing the
John Joseph wrote:
Was able to communicate clearly with e60 and E61
with asterisk with new access point
[..]
Could you please post some details (or even better: write them in some sort
of Wiki) on the configuration you did on the Nokia?
I'm thinking about buying a Nokia E60 but after a
Colin Anderson wrote:
C'mon guys! Certify a few current model servers and be done
with it.
Problem is, certification is a moving target and can become
invalid with something as simple as a BIOS change by the
manufacturer. Now that the barrier to entry to changing
On Jun 11, 2006, at 8:15 AM, James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I
changed
the port from 5060 to 5070 and now things are working, so something
is
definitely playing up on port 5060.
If you are behind a NAT perhaps two SIP devices are both
Guys, is there a way to set CDR vards like SRC, I tried using set but
asterisk complains they are RO vars. What Im trying to do is a small way to
let users make calls from someone elses extension but auth using a password
and seitch credential to their own so the call appears on CDR as made from
I'm looking at setting up an ISDN internet service for someone, and
she'd like to be able to do VoIP. The modem (230kbps serial and 2 POTS
ports) you get from the ISP can do DVO (Dynamic Voice Override) where
you can be online at 128kbits/sec (2 channels), but if a voice call is
detected (call
James Harper wrote:
Additionally, just to satisfy myself that I wasn't going mad I
changed
the port from 5060 to 5070 and now things are working, so something
is
definitely playing up on port 5060.
James
You probably have are behind NAT and your NAT device has a SIP ALG.
Changing
HI, Anyone knows the current status of JIAXclient? I tried to recompile the sources available in sourceforge but they reference a old java package that I was not able to find. I tried to e-mail the author but seems that his account is no longer valid.
I in need of a java IAX client that could be
Rubens Zupelli Filho [EMAIL PROTECTED] writes:
Anyone knows the current status of JIAXclient?
I have been playing with jiaxclient 0.0.6, and it seems to mostly work
if you have a working copy of the C iaxclient library. I would test
iaxclient with the command-line tools that come with it and
Scott,
You are compiling in Linux or Windows?
The package the java compiler is not founding is:
net.sourceforge.iaxclient.jni
many thanks.
On 6/11/06, Scott Gifford [EMAIL PROTECTED] wrote:
Rubens Zupelli Filho [EMAIL PROTECTED] writes:
Anyone knows the current status of JIAXclient?
I
I will be out of the office starting 12/06/2006 and will not return until
17/06/2006.
Dear Sir / Mdm,
I'm currently travelling.
During this period of time, I have minimal access to internet and email. As
such, please be aware that I might not be able to reply to your queries
promptly. I
Not being very happy with festival I would like ro get a better TTS
engine. I looked at the listings at:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international
but I would like to get user input on suggested packages for Linux. Best
performance vs. cost
Doug
I'm having the exact same problem. Please any ideas? My IP phones keep ringing after PSTN hangup or PSTN answer... for about 6 or 7 seconds.On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show
i guess you were right.
it was due to the previous version of asterisk on my PC,although i had make clean it
anyway thanx for the help.
can you tell me if i can use the same iax.conf and extensions.conf files that i used for asterisk-1.0.3 for this 1.2.9.1 version?
thanx again
On 6/11/06, Thomas
Hi Amna, Can use all the archives * conf.
In this case you will be making one upgrade of version of asterisk.
I wait to have helped.
Best RegardsJosué
2006/6/12, amna saleem [EMAIL PROTECTED]:
i guess you were right.
it was due to the previous version of asterisk on my PC,although i had make
Hi Issac,
Ok, here goes :) Again, my disclaimer-- I'm pretty new to Asterisk, so
I'm sure half of this is not needed or potentially even misconfigured.
You will even see some lines commented out, since I wanted to test if
they were needed--they weren't. I'm hoping to clean everything up and
put
Rubens Zupelli Filho [EMAIL PROTECTED] writes:
You are compiling in Linux or Windows?
Both. It works on Linux, but not yet on Windows.
The package the java compiler is not founding is:
net.sourceforge.iaxclient.jni
That's part of the source package; probably the classpath just needs
to be
Hi,
Im an unsuccessful user of E60. Please post the configs on the phone in detail
Thanks
Dan
On 11/06/06, Markus Schuster [EMAIL PROTECTED] wrote:
John Joseph wrote:
Was able to communicate clearly with e60 and E61
with asterisk with new access point
[..]
Could you please post
Hy men, :-)
Use Industrial PICMG PC's.
Higher cost at buy, but very stable and evolutive platforms.
SBC doesn't change during a long industrial period.
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve
Thanx for everyone's passionate responses, and apologises for not
replying sooner.
1. Based on what I have seen I take it noone is sure of what the true
purpose and the effects of the relaxdtmf parameter offer.
2. I am using both a mixture of VSP's and SPA3K's, but primarily it
is the
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